similar to: Forwarded call flag

Hi, In our office, we're slowly migrating from a cisco call manager set up to asterisk. Problem is management doesn't want to buy any other hardware ?as they had already invested a lot in cisco. The main cause of this is asterisk's added features like unique FAX number for everyone in the company (which will be the same as phone DID), Voice mail, Auto Answer etc yet we need thousands

I there, I have some trouble to do working caller id name for outgoing calls on the PRI we just installed. Caller id name work on incoming calls only. Caller id number work on incoming and outgoing calls. The provider, Goup Telecom, said that's in what i'm sending. They said that I send the cid name in ascii code and to do it working, I need to send it in hex. So I take some traces

Hi! Has someone encountered the following problem? What am I doing wrong or does someone know a workaround? Dovecot 1.2.1 on Debian unstable, two namespaces: namespace private { prefix = location = maildir:~/res/Maildir:LAYOUT=fs inbox = yes list = yes } namespace private { prefix = gtd/ separator = / list = yes location =

I have the following situation.... I'm using Action Originate to originate a call for a costumer. Originate goes to a context that call the dial application. Before the application (Dial using the G option) to be invoked i'm setting the variable cellphone like this: [firstcontext] exten => s,1,Set(CDR(cellphone)=${CELLPHONE}) exten => s,n,Dial(IAX2/user:pass at

Okay, now I am really confused. I have two PRI's coming in from two different Carriers (QWEST and ELI), both of them are supposed to be setup to pass name and number on incoming calls. Problem that I am having is that I am not receiving inbound caller id name on either PRI, the only thing that both carriers have in common is that I am terminating into a DMS switch at the carrier.

Hi Guys, i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way: 1) use a phone in PBX1 2) call extension in PBX2 3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a cellphone) my questions now is : am i gonna be able to dial from an IPphone registered within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2? anybody know

> Now I have the problem for my cellphone... I need to register from almost any > IP (at least in Europe), so I can't restrict it. > Well, the password is NOT simple and random. > > Now, I tried to register the user of my cellphone using a PC, as my cellphone > was already registered. > And Asterisk accepted this registration... :( Were you trying to register the PC

Hello, Avaya has a nice feature that allows you to a) ring both a cellphone and a desktop phone at the same time b) transfer calls (and access other PBX features) from the cellphone that recieved the call, as long as the call is bridged through the PBX c) while talking on the cellphone, pick up the handset on your desktop phone and the call is automatically moved there, hanging up the cellphone

Hi list! I'm working hard to securing my Asterisk... Now I deleted all possibility to access the node as "anonymous" and every call through the proxy will be checked (just known peers are allowed to use it). Furthermore, I restricted the registration of my home phones to the Network I reserved for them and I changed the port on my Firewall, so that I don't use 5060 anymore. Now

Hi; I've been using Asterisk for a few months now, and I have run into an interesting issue that I thought someone else in the community may have run into: I have an Asterisk install set up to receive helpdesk calls, route them to several IAX extensions and an extension which is simply a forwarded call over the POTS to a cellphone, so that if no one is logged into their IAX extensions for

Hello I don't have a landine and use a VOSP to provide access to the telephone network. In case a call comes in and I'm not home, I'd like Asterisk to log the call, and then send an SIP message to my VOSP so the call is forwarded to my cellphone and is thus charged to the caller, without Asterisk having to dial out to my cellphone through my VOSP at my expense and bridge the two

Zitat von Steve Totaro <stotaro at totarotechnologies.com>: > Are you using the wifi on on the cellphone? The peer IP is showing as > 192.168.200.3 which is not a routable address. Unless things have changed, > double NAT configurations do not work. Hi Steve, My Asterisk is behind a NAT, but my cellphone was NOT in NAT, but direct in Internet. But maybe my Provider does a

I should've mentioned that before. I've tried doing that and it has no effect. I've tried both upper and lower-case 'r's. I've also tried a workaround that I thought would work, but it doesn't: Answering the call and then using the playtones(ringing) command before connecting to my cellphone. -----Original Message----- Date: Sat, 1 Apr 2006 19:59:46 +0100 From:

Hi again, list! I know, I'm really annoying the list... :) Well, maybe I got my Asterisk at home ("wrt" on the previous E-Mails) to accept my mobile phone from Internet. It was a problem with the network and the firewall. Now I can log my mobile phone in my Asterisk in and the phone is REACHABLE. Wow! Got it! If I call a phone at home using my cellphone it works and the

Hello, I am looking to purchase a new quad-band cellphone and I'm looking for one with WiFi and enough CPU power for stable SIP calls. I was wondering if anyone here can share his experience and recommend on a good cellphone. Any chance there is such a phone with even good WiFi profiles management or am I asking for too much now? :-) Thanks, Tomer.

Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo cancelation and a quad FXO card. We have 4 analog lines, one of which is a Cellphone line for least cost

Dear R cognoscenti, While having NA as a native type is nifty, it is annoying when making binary choices. Question: Is there anything bad about writing comparison functions that behavior like %in% (which I love) and ignore NAs? "%>%" <- function(table, x) { return(which(table > x)) } "%<%" <- function(table, x) { return(which(table < x)) } test <-

Hello, Are there any gateways which allow me to hook a cellphone to Asterisk and use that line for routing my calls? Basically, I'm looking to play around a bit and if I can get to connect a cellphone with Asterisk then that would be great. Thanks, Hitesh PS: I have tried to search on the web, but didn't find any pointers on how to do so. -------------- next part -------------- An HTML

Hello Everyone. I usually find my own solutions for problems but this time, after several months, I've given up. My asterisk is set up so that incoming calls from my voip provider ring on both my sip extension and my cellphone at the same time. When the system receives an incoming call, ringtones indicating that the call is being connected play normally for the first 5 seconds to the

Hi, Here's the deal. When someone leaves me a voicemail message I want Asterisk to call me on my cellphone by dialing my cellphone number and tell me I have a message. Is this possible? Can anyone cite examples? Most commercial voicemail systems produced in the last 10 years can do this. Any help would be much appreciated. Regards, Mike -------------- next part -------------- An HTML